facing problem with originating webrtc calls
1-when iam doing call from webrtc iget ice working
<--- SIP read from WS:91.196.158.205:1466 --->
INVITE sip:0669197533 at 77.91.132.9 SIP/2.0
Via: SIP/2.0/WS 7cvtd9ihs2e8.invalid;branch=z9hG4bK8749315
Max-Forwards: 69
To: <sip:0669197533 at 77.91.132.9>
From: "Anton" <sip:1065 at 77.91.132.9>;tag=5i21qaop43
Call-ID: ocq4hu8eol3kijsgvt6b
CSeq: 1465 INVITE
Authorization: Digest algorithm=MD5, username="1065",
realm="77.91.132.9",
nonce="5152b137", uri="sip:0669197533 at 77.91.132.9",
response="446883f3c97a49ea7a9a554a1ba31b6a"
X-Can-Renegotiate: true
Contact: <sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS
Supported: timer,ice,outbound
User-Agent: JsSIP 0.6.26
Content-Length: 2554
v=0
o=- 4785391175048354014 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
m=audio 2313 RTP/SAVPF 111 103 104 9 0 8 106 105 13 126
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:3696151487 cname:jXfPZ33h32Mx9liw
a=ssrc:3696151487 msid:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
8a2acec3-8511-4d36-9b51-05b8752c2ddd
a=ssrc:3696151487 mslabel:cC3clldcCIxZyWm8eHpKdycUakfANCZmV8Br
a=ssrc:3696151487 label:8a2acec3-8511-4d36-9b51-05b8752c2ddd
m=video 2313 RTP/SAVPF 100 116 117 96
c=IN IP4 192.168.88.26
a=rtcp:2313 IN IP4 192.168.88.26
a=candidate:97470069 1 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:97470069 2 udp 2122260223 192.168.88.26 2313 typ host
generation 0
a=candidate:1263319685 1 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=recvonly
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
2-BUT when i do channel originate sip/GOROD/XXXXX extension 1065 at office
-- Executing [1065 at office:1] Dial("SIP/GOROD-00000004",
"SIP/1065") in
new stack
== Using SIP RTP CoS mark 5
[Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("7cvtd9ihs2e8.invalid",
"(null)", ...):
Name or service not known
[Apr 28 14:07:47] WARNING[4006][C-00000032]: chan_sip.c:15869
__set_address_from_contact: Invalid host name in Contact: (can't resolve in
DNS) : '7cvtd9ihs2e8.invalid'
[Apr 28 14:07:47] ERROR[4006][C-00000032]: netsock2.c:98
ast_sockaddr_stringify_fmt: getnameinfo(): ai_family not supported
Audio is at 16476
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 91.196.158.205:1466:
INVITE sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 77.91.132.9:5060;branch=z9hG4bK3f293d79;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk at 77.91.132.9>;tag=as78119d2b
To: <sip:0momhddj at 7cvtd9ihs2e8.invalid;transport=ws>
Contact: <sip:asterisk at 77.91.132.9:5060;transport=WS>
Call-ID: 17a96e0848cdd7d226d3665a36c65c77 at 77.91.132.9:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.15.0
Date: Tue, 28 Apr 2015 11:07:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 437
v=0
o=root 1122885298 1122885298 IN IP4 77.91.132.9
s=Asterisk PBX 11.15.0
c=IN IP4 77.91.132.9
t=0 0
m=audio 16476 RTP/SAVPF 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CC:82:C8:04:1F:DC:FE:B7:56:27:26:FF:18:CD:BB:71:99:B8:97:F9:81:2B:08:74:72:67:3B:A9:88:5F:00:34
a=sendrecv
thats why i got Failed to set remote offer sdp: Called with SDP without
ice-ufrag and ice-pwd
Waiting for your advice ---thanks
--
Best regards
Antony
??? (066) 919-75-33
??? (063) 656-43-40
satskiy.a at gmail.com <mail%3Asatskiy.a at gmail.com>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150428/4f152d16/attachment.html>
